1. Field of the Invention
The present invention relates to the field of telecommunications and, in particular, to a method and apparatus for facilitating tiered collaboration.
2. Background Information
Numerous advances have been made in recent years in the field of telecommunications. One example of these numerous advances in communications is the emerging field of computer telephony via the internet. In particular, the field of internet telephony has emerged as a viable technology that is evolving at an ever increasing rate. Evidence of this evolution of internet telephony is best characterized by the number of new products recently become available in the market. Products such as CoolTalk by Netscape Communications Corporation of Mountain View, Calif.; Internet Connection Phone by International Business Machines of Amonk, N.Y.; Intel Internet Phone (IPhone) by Intel Corporation of Santa Clara, Calif.; NetMeeting by Microsoft Corporation, Redmond, Wash.; Quarterdeck WebTalk by Quarterdeck Corporation of Marina Del Rey, Calif.; TeleVox by Voxware Incorporated of Princeton, N.J.; and WebPhone by Netspeak Corporation of Boca Raton, Fla., are representative of the current state of applications facilitating interent telephony.
Each of these products offers internet based voice communications with a telephone motif, between two users each using the same (or compatible) product on either end of the internet connection. That is, the internet provides the "switching" architecture for the communication system, while the computer acts as the audio interface (e.g., the "handset"). One reason for the proliferation of these applications is a desire to push the technology of the internet to provide a total communications tool. The appeal to users is that, currently, the use of the internet is free of toll charges. Therefore, currently, a user of an internet phone product may communicate with another user located anywhere else in the world without having to pay the long distance charges associated with making a telephone call using the public switched telephone network (PSTN), so long as each of the users has a computer that is appropriately configured to provide such communications.
Although innovative in their own right, the current internet based telephony applications identified above have a number of limitations which retard their acceptance as a primary communications tool. One such limitation is that many of the applications identified above require that both users have installed the same software package, or compatible packages and, therefore, provide a relatively low level of interoperability. One reason for this lack of interoperability between internet telephony applications is that the developers of many of these products have incorporated different voice encoders (commonly referred to as a "voice codec", or simply a "codec" by those in the telecommunication arts) in the products. Consequently, as a result of the different codecs used, many internet telephony applications are unable to recognize speech encoded (i.e., digitized) by a codec of a disimilar application.
This problem is alleviated for those products that are upgraded to comply with emerging telephony standards, such as International Telecommunication Union's (ITU) H.323 standard. However, other limitations remain. For example, another limitation associated with many of these products is that they are tied to the internet, often requiring all users to access a common server in order to maintain a directory of available users in which to call. That is to say, many of the applications identified above do not integrate the packet switched network of the internet with the circuit switched public switched telephone network (PSTN). Therefore, although a computer connected to the internet may communicate with another user on the internet, assuming they are both using a common software application (or at least applications with compatible codecs), these applications do not support communication with a user of a Telephone handset.
The reason for this limitation is readily understood by those who appreciate the complexity of the two networks. As alluded to above, the internet is a packet switched network. That is to say, communication over the internet is accomplished by "breaking" the transmitted data into varying-sized packages (or "packets"), based primarily on communication content, and interleaving the various-sized packages to best utilize the bandwidth available at any given time on the internet. When the packets reach their intended destination, they must be reassembled into the originally transmitted data. Loss of packets, and thus data, occur frequently in such a network, and the ability of the network to successfully transmit information from one point in the network to another determines the quality of the network. For inter-computer communication transactions involving non real-time data, the ability to transmit packets and retransmit any packets that are perceived to have been dropped is not a severe limitation and may not even be perceived by the user of the system. However, in a voice communication transaction, the delay required to retransmit even one data packet may be perceived by a user. At best, such delays are an annoying inconvenience. In practice, the delays actually can become intolerable, as the cumulative latency adds up with successive transmissions.
In contrast to the packet switched network of the internet, the public switched telephone network (PSTN) is a circuit switched network. That is to say that the PSTN assigns a dedicated communication line to a user with which to complete the telephone call, wherein the user can utilize the assigned resource of the PSTN in any way they choose, with the understanding that the user is paying for the use of the dedicated resource of the PSTN. While the circuit switched approach of the PSTN system is not necessarily the most efficient system in terms of call traffic (i.e., it does not make use of the "dead space" common in a conversation), it is relatively easy to ensure that information destined for a particular user is delivered, it simply provides a dedicated line to complete the transaction.
Nonetheless, despite these engineering challanges, a few products have emerged which purport to integrate the PSTN to the internet. Products such as Net2Phone by IDT Corporation of Hackensack, N.J., claim to provide a computer user with the ability to place and receive a phone call to/from a PSTN extension. Unfortunately, none of these products completely solve the problem of integrating the two networks. The limitations perhaps best characterized by way of an example communication session. With these prior art internet telephony applications, a user of an internet telephony enabled client computer initiating a telephone call to a Telephone handset launches the collaboration session from the client computer by accessing a server (the primary access server), operated by the developer of the internet telephony application that supports internet telecommunications. As the initiator accesses the primary access server, he/she is prompted for a destination address, which takes the form of a PSTN telephone number for an outgoing call to a Telephone handset. Having provided the primary access server with the PSTN telephone number associated with the Telephone handset, the primary server somehow determines.sup.1 which server in a community of similarly enabled servers (i.e., servers with the hardware/software necessary to provide access to the PSTN) is closest to the destination address, and completes the telephone call by routing the telephone call through a number of intermediate servers on the internet to the selected server, which will actually place the collaboration session to the Telephone handset on behalf of the client computer, facilitating the collaboration session between the client computer and the Telephone handset. In other words, the user of the client computer is required to have prior knowledge of the destination phone number, which is limiting in many circumstances. For example, in a situation where the user of the client computer is engaged in a data communication session involving a webpage for a corporate entity, the user may wish to speak with someone in a "local office" of the corporate entity. Prior art internet telephony applications require that the telephone number for the "local office" of the corporate entity be provided by the user of the client computer in order to place the telephone call. If the telephone number for the "local office" of the corporate entity is not provided by the webpage, the user of client computer must look it up or have prior knowledge of it.
 FNT .sup.1 The manner in which the "primary access server" determines the "call originating server" is not known.
Additionally, while the prior art approach of simply finding the internet telephony enabled server closest to the destination address may offer the simplest technical solution and a seemingly cheaper connection, it does not ensure the quality of the voice connection. One skilled in the art will appreciate that there are a number of characteristics which may impact the quality of the voice connection. For example, insofar as the internet is a packet switched network, as the number of intermediate routers required to interface the client computer to the selected server increases so, too, does the likelihood that data packets containing voice information could be lost or corrupted. The result of lost or corrupted data packets is broken or garbled speech. Another factor affecting internet telephony communication performance is the bandwidth available on the selected server. If, for example, the selected server is very busy handling a number of other processes, the performance associated with each of the processes begins to degrade (i.e., slow down), which may also result in delayed delivery of data packets containing speech, which in turn results in user perception of poor quality.
Those skilled in the art will appreciate that in order to take advantage of prior art internet telephony systems, a client computer system must be appropriately endowed with the proper input/output (I/O) components (sometimes referred to as "peripherals"). Speakers, an audio sound board and a microphone are but a few examples of the components necessary to place and receive audio phone calls over the internet using ones computer. Many of the newer computer's sold today have this hardware pre-installed as a part of a system "bundle". However, many of the low-end entry level systems may not have audio/video (a/v) input/output (I/O) equipment included in its system "bundle", in an effort to keep the cost of the system down. Moreover, a number of the older computers did not have audio/video components included in the system when sold, so unless the end-user has since added that equipment, many of the older computers in service may not have the equipment necessary to take advantage of the prior art internet telephony services.
Thus, a need exists for a method and apparatus for facilitating tiered collaboration commensurate with the system attributes of the client computer that is unencumbered by the limitations associated with the prior art.